7dede4c87594ed2138fbb492fed223dbaf432a05
[libav.git] / libavformat / rtsp.c
1 /*
2 * RTSP/SDP client
3 * Copyright (c) 2002 Fabrice Bellard
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 #include "libavutil/base64.h"
23 #include "libavutil/avstring.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/random_seed.h"
26 #include "avformat.h"
27
28 #include <sys/time.h>
29 #if HAVE_SYS_SELECT_H
30 #include <sys/select.h>
31 #endif
32 #include <strings.h>
33 #include "internal.h"
34 #include "network.h"
35 #include "os_support.h"
36 #include "http.h"
37 #include "rtsp.h"
38
39 #include "rtpdec.h"
40 #include "rdt.h"
41 #include "rtpdec_asf.h"
42
43 //#define DEBUG
44 //#define DEBUG_RTP_TCP
45
46 #if LIBAVFORMAT_VERSION_INT < (53 << 16)
47 int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP);
48 #endif
49
50 /* Timeout values for socket select, in ms,
51 * and read_packet(), in seconds */
52 #define SELECT_TIMEOUT_MS 100
53 #define READ_PACKET_TIMEOUT_S 10
54 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_TIMEOUT_MS
55
56 static void get_word_until_chars(char *buf, int buf_size,
57 const char *sep, const char **pp)
58 {
59 const char *p;
60 char *q;
61
62 p = *pp;
63 p += strspn(p, SPACE_CHARS);
64 q = buf;
65 while (!strchr(sep, *p) && *p != '\0') {
66 if ((q - buf) < buf_size - 1)
67 *q++ = *p;
68 p++;
69 }
70 if (buf_size > 0)
71 *q = '\0';
72 *pp = p;
73 }
74
75 static void get_word_sep(char *buf, int buf_size, const char *sep,
76 const char **pp)
77 {
78 if (**pp == '/') (*pp)++;
79 get_word_until_chars(buf, buf_size, sep, pp);
80 }
81
82 static void get_word(char *buf, int buf_size, const char **pp)
83 {
84 get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
85 }
86
87 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
88 static int sdp_parse_rtpmap(AVFormatContext *s,
89 AVCodecContext *codec, RTSPStream *rtsp_st,
90 int payload_type, const char *p)
91 {
92 char buf[256];
93 int i;
94 AVCodec *c;
95 const char *c_name;
96
97 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
98 * see if we can handle this kind of payload.
99 * The space should normally not be there but some Real streams or
100 * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
101 * have a trailing space. */
102 get_word_sep(buf, sizeof(buf), "/ ", &p);
103 if (payload_type >= RTP_PT_PRIVATE) {
104 RTPDynamicProtocolHandler *handler;
105 for (handler = RTPFirstDynamicPayloadHandler;
106 handler; handler = handler->next) {
107 if (!strcasecmp(buf, handler->enc_name) &&
108 codec->codec_type == handler->codec_type) {
109 codec->codec_id = handler->codec_id;
110 rtsp_st->dynamic_handler = handler;
111 if (handler->open)
112 rtsp_st->dynamic_protocol_context = handler->open();
113 break;
114 }
115 }
116 } else {
117 /* We are in a standard case
118 * (from http://www.iana.org/assignments/rtp-parameters). */
119 /* search into AVRtpPayloadTypes[] */
120 codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
121 }
122
123 c = avcodec_find_decoder(codec->codec_id);
124 if (c && c->name)
125 c_name = c->name;
126 else
127 c_name = "(null)";
128
129 get_word_sep(buf, sizeof(buf), "/", &p);
130 i = atoi(buf);
131 switch (codec->codec_type) {
132 case AVMEDIA_TYPE_AUDIO:
133 av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);
134 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
135 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
136 if (i > 0) {
137 codec->sample_rate = i;
138 get_word_sep(buf, sizeof(buf), "/", &p);
139 i = atoi(buf);
140 if (i > 0)
141 codec->channels = i;
142 // TODO: there is a bug here; if it is a mono stream, and
143 // less than 22000Hz, faad upconverts to stereo and twice
144 // the frequency. No problem, but the sample rate is being
145 // set here by the sdp line. Patch on its way. (rdm)
146 }
147 av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
148 codec->sample_rate);
149 av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",
150 codec->channels);
151 break;
152 case AVMEDIA_TYPE_VIDEO:
153 av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);
154 break;
155 default:
156 break;
157 }
158 return 0;
159 }
160
161 /* parse the attribute line from the fmtp a line of an sdp response. This
162 * is broken out as a function because it is used in rtp_h264.c, which is
163 * forthcoming. */
164 int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
165 char *value, int value_size)
166 {
167 *p += strspn(*p, SPACE_CHARS);
168 if (**p) {
169 get_word_sep(attr, attr_size, "=", p);
170 if (**p == '=')
171 (*p)++;
172 get_word_sep(value, value_size, ";", p);
173 if (**p == ';')
174 (*p)++;
175 return 1;
176 }
177 return 0;
178 }
179
180 /** Parse a string p in the form of Range:npt=xx-xx, and determine the start
181 * and end time.
182 * Used for seeking in the rtp stream.
183 */
184 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
185 {
186 char buf[256];
187
188 p += strspn(p, SPACE_CHARS);
189 if (!av_stristart(p, "npt=", &p))
190 return;
191
192 *start = AV_NOPTS_VALUE;
193 *end = AV_NOPTS_VALUE;
194
195 get_word_sep(buf, sizeof(buf), "-", &p);
196 *start = parse_date(buf, 1);
197 if (*p == '-') {
198 p++;
199 get_word_sep(buf, sizeof(buf), "-", &p);
200 *end = parse_date(buf, 1);
201 }
202 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
203 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
204 }
205
206 typedef struct SDPParseState {
207 /* SDP only */
208 struct in_addr default_ip;
209 int default_ttl;
210 int skip_media; ///< set if an unknown m= line occurs
211 } SDPParseState;
212
213 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
214 int letter, const char *buf)
215 {
216 RTSPState *rt = s->priv_data;
217 char buf1[64], st_type[64];
218 const char *p;
219 enum AVMediaType codec_type;
220 int payload_type, i;
221 AVStream *st;
222 RTSPStream *rtsp_st;
223 struct in_addr sdp_ip;
224 int ttl;
225
226 dprintf(s, "sdp: %c='%s'\n", letter, buf);
227
228 p = buf;
229 if (s1->skip_media && letter != 'm')
230 return;
231 switch (letter) {
232 case 'c':
233 get_word(buf1, sizeof(buf1), &p);
234 if (strcmp(buf1, "IN") != 0)
235 return;
236 get_word(buf1, sizeof(buf1), &p);
237 if (strcmp(buf1, "IP4") != 0)
238 return;
239 get_word_sep(buf1, sizeof(buf1), "/", &p);
240 if (ff_inet_aton(buf1, &sdp_ip) == 0)
241 return;
242 ttl = 16;
243 if (*p == '/') {
244 p++;
245 get_word_sep(buf1, sizeof(buf1), "/", &p);
246 ttl = atoi(buf1);
247 }
248 if (s->nb_streams == 0) {
249 s1->default_ip = sdp_ip;
250 s1->default_ttl = ttl;
251 } else {
252 st = s->streams[s->nb_streams - 1];
253 rtsp_st = st->priv_data;
254 rtsp_st->sdp_ip = sdp_ip;
255 rtsp_st->sdp_ttl = ttl;
256 }
257 break;
258 case 's':
259 av_metadata_set2(&s->metadata, "title", p, 0);
260 break;
261 case 'i':
262 if (s->nb_streams == 0) {
263 av_metadata_set2(&s->metadata, "comment", p, 0);
264 break;
265 }
266 break;
267 case 'm':
268 /* new stream */
269 s1->skip_media = 0;
270 get_word(st_type, sizeof(st_type), &p);
271 if (!strcmp(st_type, "audio")) {
272 codec_type = AVMEDIA_TYPE_AUDIO;
273 } else if (!strcmp(st_type, "video")) {
274 codec_type = AVMEDIA_TYPE_VIDEO;
275 } else if (!strcmp(st_type, "application")) {
276 codec_type = AVMEDIA_TYPE_DATA;
277 } else {
278 s1->skip_media = 1;
279 return;
280 }
281 rtsp_st = av_mallocz(sizeof(RTSPStream));
282 if (!rtsp_st)
283 return;
284 rtsp_st->stream_index = -1;
285 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
286
287 rtsp_st->sdp_ip = s1->default_ip;
288 rtsp_st->sdp_ttl = s1->default_ttl;
289
290 get_word(buf1, sizeof(buf1), &p); /* port */
291 rtsp_st->sdp_port = atoi(buf1);
292
293 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
294
295 /* XXX: handle list of formats */
296 get_word(buf1, sizeof(buf1), &p); /* format list */
297 rtsp_st->sdp_payload_type = atoi(buf1);
298
299 if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
300 /* no corresponding stream */
301 } else {
302 st = av_new_stream(s, 0);
303 if (!st)
304 return;
305 st->priv_data = rtsp_st;
306 rtsp_st->stream_index = st->index;
307 st->codec->codec_type = codec_type;
308 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
309 /* if standard payload type, we can find the codec right now */
310 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
311 }
312 }
313 /* put a default control url */
314 av_strlcpy(rtsp_st->control_url, rt->control_uri,
315 sizeof(rtsp_st->control_url));
316 break;
317 case 'a':
318 if (av_strstart(p, "control:", &p)) {
319 if (s->nb_streams == 0) {
320 if (!strncmp(p, "rtsp://", 7))
321 av_strlcpy(rt->control_uri, p,
322 sizeof(rt->control_uri));
323 } else {
324 char proto[32];
325 /* get the control url */
326 st = s->streams[s->nb_streams - 1];
327 rtsp_st = st->priv_data;
328
329 /* XXX: may need to add full url resolution */
330 av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
331 NULL, NULL, 0, p);
332 if (proto[0] == '\0') {
333 /* relative control URL */
334 if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
335 av_strlcat(rtsp_st->control_url, "/",
336 sizeof(rtsp_st->control_url));
337 av_strlcat(rtsp_st->control_url, p,
338 sizeof(rtsp_st->control_url));
339 } else
340 av_strlcpy(rtsp_st->control_url, p,
341 sizeof(rtsp_st->control_url));
342 }
343 } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
344 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
345 get_word(buf1, sizeof(buf1), &p);
346 payload_type = atoi(buf1);
347 st = s->streams[s->nb_streams - 1];
348 rtsp_st = st->priv_data;
349 sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p);
350 } else if (av_strstart(p, "fmtp:", &p) ||
351 av_strstart(p, "framesize:", &p)) {
352 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
353 // let dynamic protocol handlers have a stab at the line.
354 get_word(buf1, sizeof(buf1), &p);
355 payload_type = atoi(buf1);
356 for (i = 0; i < s->nb_streams; i++) {
357 st = s->streams[i];
358 rtsp_st = st->priv_data;
359 if (rtsp_st->sdp_payload_type == payload_type &&
360 rtsp_st->dynamic_handler &&
361 rtsp_st->dynamic_handler->parse_sdp_a_line)
362 rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,
363 rtsp_st->dynamic_protocol_context, buf);
364 }
365 } else if (av_strstart(p, "range:", &p)) {
366 int64_t start, end;
367
368 // this is so that seeking on a streamed file can work.
369 rtsp_parse_range_npt(p, &start, &end);
370 s->start_time = start;
371 /* AV_NOPTS_VALUE means live broadcast (and can't seek) */
372 s->duration = (end == AV_NOPTS_VALUE) ?
373 AV_NOPTS_VALUE : end - start;
374 } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
375 if (atoi(p) == 1)
376 rt->transport = RTSP_TRANSPORT_RDT;
377 } else {
378 if (rt->server_type == RTSP_SERVER_WMS)
379 ff_wms_parse_sdp_a_line(s, p);
380 if (s->nb_streams > 0) {
381 if (rt->server_type == RTSP_SERVER_REAL)
382 ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
383
384 rtsp_st = s->streams[s->nb_streams - 1]->priv_data;
385 if (rtsp_st->dynamic_handler &&
386 rtsp_st->dynamic_handler->parse_sdp_a_line)
387 rtsp_st->dynamic_handler->parse_sdp_a_line(s,
388 s->nb_streams - 1,
389 rtsp_st->dynamic_protocol_context, buf);
390 }
391 }
392 break;
393 }
394 }
395
396 static int sdp_parse(AVFormatContext *s, const char *content)
397 {
398 const char *p;
399 int letter;
400 /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,
401 * contain long SDP lines containing complete ASF Headers (several
402 * kB) or arrays of MDPR (RM stream descriptor) headers plus
403 * "rulebooks" describing their properties. Therefore, the SDP line
404 * buffer is large.
405 *
406 * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line
407 * in rtpdec_xiph.c. */
408 char buf[16384], *q;
409 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
410
411 memset(s1, 0, sizeof(SDPParseState));
412 p = content;
413 for (;;) {
414 p += strspn(p, SPACE_CHARS);
415 letter = *p;
416 if (letter == '\0')
417 break;
418 p++;
419 if (*p != '=')
420 goto next_line;
421 p++;
422 /* get the content */
423 q = buf;
424 while (*p != '\n' && *p != '\r' && *p != '\0') {
425 if ((q - buf) < sizeof(buf) - 1)
426 *q++ = *p;
427 p++;
428 }
429 *q = '\0';
430 sdp_parse_line(s, s1, letter, buf);
431 next_line:
432 while (*p != '\n' && *p != '\0')
433 p++;
434 if (*p == '\n')
435 p++;
436 }
437 return 0;
438 }
439
440 /* close and free RTSP streams */
441 void ff_rtsp_close_streams(AVFormatContext *s)
442 {
443 RTSPState *rt = s->priv_data;
444 int i;
445 RTSPStream *rtsp_st;
446
447 for (i = 0; i < rt->nb_rtsp_streams; i++) {
448 rtsp_st = rt->rtsp_streams[i];
449 if (rtsp_st) {
450 if (rtsp_st->transport_priv) {
451 if (s->oformat) {
452 AVFormatContext *rtpctx = rtsp_st->transport_priv;
453 av_write_trailer(rtpctx);
454 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
455 uint8_t *ptr;
456 url_close_dyn_buf(rtpctx->pb, &ptr);
457 av_free(ptr);
458 } else {
459 url_fclose(rtpctx->pb);
460 }
461 av_metadata_free(&rtpctx->streams[0]->metadata);
462 av_metadata_free(&rtpctx->metadata);
463 av_free(rtpctx->streams[0]);
464 av_free(rtpctx);
465 } else if (rt->transport == RTSP_TRANSPORT_RDT)
466 ff_rdt_parse_close(rtsp_st->transport_priv);
467 else
468 rtp_parse_close(rtsp_st->transport_priv);
469 }
470 if (rtsp_st->rtp_handle)
471 url_close(rtsp_st->rtp_handle);
472 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
473 rtsp_st->dynamic_handler->close(
474 rtsp_st->dynamic_protocol_context);
475 }
476 }
477 av_free(rt->rtsp_streams);
478 if (rt->asf_ctx) {
479 av_close_input_stream (rt->asf_ctx);
480 rt->asf_ctx = NULL;
481 }
482 }
483
484 static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st,
485 URLContext *handle)
486 {
487 RTSPState *rt = s->priv_data;
488 AVFormatContext *rtpctx;
489 int ret;
490 AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
491
492 if (!rtp_format)
493 return NULL;
494
495 /* Allocate an AVFormatContext for each output stream */
496 rtpctx = avformat_alloc_context();
497 if (!rtpctx)
498 return NULL;
499
500 rtpctx->oformat = rtp_format;
501 if (!av_new_stream(rtpctx, 0)) {
502 av_free(rtpctx);
503 return NULL;
504 }
505 /* Copy the max delay setting; the rtp muxer reads this. */
506 rtpctx->max_delay = s->max_delay;
507 /* Copy other stream parameters. */
508 rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;
509
510 /* Set the synchronized start time. */
511 rtpctx->start_time_realtime = rt->start_time;
512
513 /* Remove the local codec, link to the original codec
514 * context instead, to give the rtp muxer access to
515 * codec parameters. */
516 av_free(rtpctx->streams[0]->codec);
517 rtpctx->streams[0]->codec = st->codec;
518
519 if (handle) {
520 url_fdopen(&rtpctx->pb, handle);
521 } else
522 url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
523 ret = av_write_header(rtpctx);
524
525 if (ret) {
526 if (handle) {
527 url_fclose(rtpctx->pb);
528 } else {
529 uint8_t *ptr;
530 url_close_dyn_buf(rtpctx->pb, &ptr);
531 av_free(ptr);
532 }
533 av_free(rtpctx->streams[0]);
534 av_free(rtpctx);
535 return NULL;
536 }
537
538 /* Copy the RTP AVStream timebase back to the original AVStream */
539 st->time_base = rtpctx->streams[0]->time_base;
540 return rtpctx;
541 }
542
543 static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
544 {
545 RTSPState *rt = s->priv_data;
546 AVStream *st = NULL;
547
548 /* open the RTP context */
549 if (rtsp_st->stream_index >= 0)
550 st = s->streams[rtsp_st->stream_index];
551 if (!st)
552 s->ctx_flags |= AVFMTCTX_NOHEADER;
553
554 if (s->oformat) {
555 rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle);
556 /* Ownership of rtp_handle is passed to the rtp mux context */
557 rtsp_st->rtp_handle = NULL;
558 } else if (rt->transport == RTSP_TRANSPORT_RDT)
559 rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
560 rtsp_st->dynamic_protocol_context,
561 rtsp_st->dynamic_handler);
562 else
563 rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,
564 rtsp_st->sdp_payload_type);
565
566 if (!rtsp_st->transport_priv) {
567 return AVERROR(ENOMEM);
568 } else if (rt->transport != RTSP_TRANSPORT_RDT) {
569 if (rtsp_st->dynamic_handler) {
570 rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
571 rtsp_st->dynamic_protocol_context,
572 rtsp_st->dynamic_handler);
573 }
574 }
575
576 return 0;
577 }
578
579 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
580 static int rtsp_probe(AVProbeData *p)
581 {
582 if (av_strstart(p->filename, "rtsp:", NULL))
583 return AVPROBE_SCORE_MAX;
584 return 0;
585 }
586
587 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
588 {
589 const char *p;
590 int v;
591
592 p = *pp;
593 p += strspn(p, SPACE_CHARS);
594 v = strtol(p, (char **)&p, 10);
595 if (*p == '-') {
596 p++;
597 *min_ptr = v;
598 v = strtol(p, (char **)&p, 10);
599 *max_ptr = v;
600 } else {
601 *min_ptr = v;
602 *max_ptr = v;
603 }
604 *pp = p;
605 }
606
607 /* XXX: only one transport specification is parsed */
608 static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
609 {
610 char transport_protocol[16];
611 char profile[16];
612 char lower_transport[16];
613 char parameter[16];
614 RTSPTransportField *th;
615 char buf[256];
616
617 reply->nb_transports = 0;
618
619 for (;;) {
620 p += strspn(p, SPACE_CHARS);
621 if (*p == '\0')
622 break;
623
624 th = &reply->transports[reply->nb_transports];
625
626 get_word_sep(transport_protocol, sizeof(transport_protocol),
627 "/", &p);
628 if (!strcasecmp (transport_protocol, "rtp")) {
629 get_word_sep(profile, sizeof(profile), "/;,", &p);
630 lower_transport[0] = '\0';
631 /* rtp/avp/<protocol> */
632 if (*p == '/') {
633 get_word_sep(lower_transport, sizeof(lower_transport),
634 ";,", &p);
635 }
636 th->transport = RTSP_TRANSPORT_RTP;
637 } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
638 !strcasecmp (transport_protocol, "x-real-rdt")) {
639 /* x-pn-tng/<protocol> */
640 get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
641 profile[0] = '\0';
642 th->transport = RTSP_TRANSPORT_RDT;
643 }
644 if (!strcasecmp(lower_transport, "TCP"))
645 th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
646 else
647 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
648
649 if (*p == ';')
650 p++;
651 /* get each parameter */
652 while (*p != '\0' && *p != ',') {
653 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
654 if (!strcmp(parameter, "port")) {
655 if (*p == '=') {
656 p++;
657 rtsp_parse_range(&th->port_min, &th->port_max, &p);
658 }
659 } else if (!strcmp(parameter, "client_port")) {
660 if (*p == '=') {
661 p++;
662 rtsp_parse_range(&th->client_port_min,
663 &th->client_port_max, &p);
664 }
665 } else if (!strcmp(parameter, "server_port")) {
666 if (*p == '=') {
667 p++;
668 rtsp_parse_range(&th->server_port_min,
669 &th->server_port_max, &p);
670 }
671 } else if (!strcmp(parameter, "interleaved")) {
672 if (*p == '=') {
673 p++;
674 rtsp_parse_range(&th->interleaved_min,
675 &th->interleaved_max, &p);
676 }
677 } else if (!strcmp(parameter, "multicast")) {
678 if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
679 th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
680 } else if (!strcmp(parameter, "ttl")) {
681 if (*p == '=') {
682 p++;
683 th->ttl = strtol(p, (char **)&p, 10);
684 }
685 } else if (!strcmp(parameter, "destination")) {
686 struct in_addr ipaddr;
687
688 if (*p == '=') {
689 p++;
690 get_word_sep(buf, sizeof(buf), ";,", &p);
691 if (ff_inet_aton(buf, &ipaddr))
692 th->destination = ntohl(ipaddr.s_addr);
693 }
694 }
695 while (*p != ';' && *p != '\0' && *p != ',')
696 p++;
697 if (*p == ';')
698 p++;
699 }
700 if (*p == ',')
701 p++;
702
703 reply->nb_transports++;
704 }
705 }
706
707 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
708 HTTPAuthState *auth_state)
709 {
710 const char *p;
711
712 /* NOTE: we do case independent match for broken servers */
713 p = buf;
714 if (av_stristart(p, "Session:", &p)) {
715 int t;
716 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
717 if (av_stristart(p, ";timeout=", &p) &&
718 (t = strtol(p, NULL, 10)) > 0) {
719 reply->timeout = t;
720 }
721 } else if (av_stristart(p, "Content-Length:", &p)) {
722 reply->content_length = strtol(p, NULL, 10);
723 } else if (av_stristart(p, "Transport:", &p)) {
724 rtsp_parse_transport(reply, p);
725 } else if (av_stristart(p, "CSeq:", &p)) {
726 reply->seq = strtol(p, NULL, 10);
727 } else if (av_stristart(p, "Range:", &p)) {
728 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
729 } else if (av_stristart(p, "RealChallenge1:", &p)) {
730 p += strspn(p, SPACE_CHARS);
731 av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
732 } else if (av_stristart(p, "Server:", &p)) {
733 p += strspn(p, SPACE_CHARS);
734 av_strlcpy(reply->server, p, sizeof(reply->server));
735 } else if (av_stristart(p, "Notice:", &p) ||
736 av_stristart(p, "X-Notice:", &p)) {
737 reply->notice = strtol(p, NULL, 10);
738 } else if (av_stristart(p, "Location:", &p)) {
739 p += strspn(p, SPACE_CHARS);
740 av_strlcpy(reply->location, p , sizeof(reply->location));
741 } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
742 p += strspn(p, SPACE_CHARS);
743 ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
744 } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
745 p += strspn(p, SPACE_CHARS);
746 ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
747 }
748 }
749
750 /* skip a RTP/TCP interleaved packet */
751 void ff_rtsp_skip_packet(AVFormatContext *s)
752 {
753 RTSPState *rt = s->priv_data;
754 int ret, len, len1;
755 uint8_t buf[1024];
756
757 ret = url_read_complete(rt->rtsp_hd, buf, 3);
758 if (ret != 3)
759 return;
760 len = AV_RB16(buf + 1);
761
762 dprintf(s, "skipping RTP packet len=%d\n", len);
763
764 /* skip payload */
765 while (len > 0) {
766 len1 = len;
767 if (len1 > sizeof(buf))
768 len1 = sizeof(buf);
769 ret = url_read_complete(rt->rtsp_hd, buf, len1);
770 if (ret != len1)
771 return;
772 len -= len1;
773 }
774 }
775
776 int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
777 unsigned char **content_ptr,
778 int return_on_interleaved_data)
779 {
780 RTSPState *rt = s->priv_data;
781 char buf[4096], buf1[1024], *q;
782 unsigned char ch;
783 const char *p;
784 int ret, content_length, line_count = 0;
785 unsigned char *content = NULL;
786
787 memset(reply, 0, sizeof(*reply));
788
789 /* parse reply (XXX: use buffers) */
790 rt->last_reply[0] = '\0';
791 for (;;) {
792 q = buf;
793 for (;;) {
794 ret = url_read_complete(rt->rtsp_hd, &ch, 1);
795 #ifdef DEBUG_RTP_TCP
796 dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
797 #endif
798 if (ret != 1)
799 return -1;
800 if (ch == '\n')
801 break;
802 if (ch == '$') {
803 /* XXX: only parse it if first char on line ? */
804 if (return_on_interleaved_data) {
805 return 1;
806 } else
807 ff_rtsp_skip_packet(s);
808 } else if (ch != '\r') {
809 if ((q - buf) < sizeof(buf) - 1)
810 *q++ = ch;
811 }
812 }
813 *q = '\0';
814
815 dprintf(s, "line='%s'\n", buf);
816
817 /* test if last line */
818 if (buf[0] == '\0')
819 break;
820 p = buf;
821 if (line_count == 0) {
822 /* get reply code */
823 get_word(buf1, sizeof(buf1), &p);
824 get_word(buf1, sizeof(buf1), &p);
825 reply->status_code = atoi(buf1);
826 } else {
827 ff_rtsp_parse_line(reply, p, &rt->auth_state);
828 av_strlcat(rt->last_reply, p, sizeof(rt->last_reply));
829 av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
830 }
831 line_count++;
832 }
833
834 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
835 av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
836
837 content_length = reply->content_length;
838 if (content_length > 0) {
839 /* leave some room for a trailing '\0' (useful for simple parsing) */
840 content = av_malloc(content_length + 1);
841 (void)url_read_complete(rt->rtsp_hd, content, content_length);
842 content[content_length] = '\0';
843 }
844 if (content_ptr)
845 *content_ptr = content;
846 else
847 av_free(content);
848
849 if (rt->seq != reply->seq) {
850 av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
851 rt->seq, reply->seq);
852 }
853
854 /* EOS */
855 if (reply->notice == 2101 /* End-of-Stream Reached */ ||
856 reply->notice == 2104 /* Start-of-Stream Reached */ ||
857 reply->notice == 2306 /* Continuous Feed Terminated */) {
858 rt->state = RTSP_STATE_IDLE;
859 } else if (reply->notice >= 4400 && reply->notice < 5500) {
860 return AVERROR(EIO); /* data or server error */
861 } else if (reply->notice == 2401 /* Ticket Expired */ ||
862 (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
863 return AVERROR(EPERM);
864
865 return 0;
866 }
867
868 int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
869 const char *method, const char *url,
870 const char *headers,
871 const unsigned char *send_content,
872 int send_content_length)
873 {
874 RTSPState *rt = s->priv_data;
875 char buf[4096], *out_buf;
876 char base64buf[AV_BASE64_SIZE(sizeof(buf))];
877
878 /* Add in RTSP headers */
879 out_buf = buf;
880 rt->seq++;
881 snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
882 if (headers)
883 av_strlcat(buf, headers, sizeof(buf));
884 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
885 if (rt->session_id[0] != '\0' && (!headers ||
886 !strstr(headers, "\nIf-Match:"))) {
887 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
888 }
889 if (rt->auth[0]) {
890 char *str = ff_http_auth_create_response(&rt->auth_state,
891 rt->auth, url, method);
892 if (str)
893 av_strlcat(buf, str, sizeof(buf));
894 av_free(str);
895 }
896 if (send_content_length > 0 && send_content)
897 av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
898 av_strlcat(buf, "\r\n", sizeof(buf));
899
900 /* base64 encode rtsp if tunneling */
901 if (rt->control_transport == RTSP_MODE_TUNNEL) {
902 av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
903 out_buf = base64buf;
904 }
905
906 dprintf(s, "Sending:\n%s--\n", buf);
907
908 url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf));
909 if (send_content_length > 0 && send_content) {
910 if (rt->control_transport == RTSP_MODE_TUNNEL) {
911 av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests "
912 "with content data not supported\n");
913 return AVERROR_PATCHWELCOME;
914 }
915 url_write(rt->rtsp_hd_out, send_content, send_content_length);
916 }
917 rt->last_cmd_time = av_gettime();
918
919 return 0;
920 }
921
922 int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
923 const char *url, const char *headers)
924 {
925 return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
926 }
927
928 int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
929 const char *headers, RTSPMessageHeader *reply,
930 unsigned char **content_ptr)
931 {
932 return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
933 content_ptr, NULL, 0);
934 }
935
936 int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
937 const char *method, const char *url,
938 const char *header,
939 RTSPMessageHeader *reply,
940 unsigned char **content_ptr,
941 const unsigned char *send_content,
942 int send_content_length)
943 {
944 RTSPState *rt = s->priv_data;
945 HTTPAuthType cur_auth_type;
946 int ret;
947
948 retry:
949 cur_auth_type = rt->auth_state.auth_type;
950 if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header,
951 send_content,
952 send_content_length)))
953 return ret;
954
955 if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0) ) < 0)
956 return ret;
957
958 if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
959 rt->auth_state.auth_type != HTTP_AUTH_NONE)
960 goto retry;
961
962 if (reply->status_code > 400){
963 av_log(s, AV_LOG_ERROR, "method %s failed, %d\n",
964 method,
965 reply->status_code);
966 av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply);
967 }
968
969 return 0;
970 }
971
972 /**
973 * @return 0 on success, <0 on error, 1 if protocol is unavailable.
974 */
975 static int make_setup_request(AVFormatContext *s, const char *host, int port,
976 int lower_transport, const char *real_challenge)
977 {
978 RTSPState *rt = s->priv_data;
979 int rtx, j, i, err, interleave = 0;
980 RTSPStream *rtsp_st;
981 RTSPMessageHeader reply1, *reply = &reply1;
982 char cmd[2048];
983 const char *trans_pref;
984
985 if (rt->transport == RTSP_TRANSPORT_RDT)
986 trans_pref = "x-pn-tng";
987 else
988 trans_pref = "RTP/AVP";
989
990 /* default timeout: 1 minute */
991 rt->timeout = 60;
992
993 /* for each stream, make the setup request */
994 /* XXX: we assume the same server is used for the control of each
995 * RTSP stream */
996
997 for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
998 char transport[2048];
999
1000 /**
1001 * WMS serves all UDP data over a single connection, the RTX, which
1002 * isn't necessarily the first in the SDP but has to be the first
1003 * to be set up, else the second/third SETUP will fail with a 461.
1004 */
1005 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&
1006 rt->server_type == RTSP_SERVER_WMS) {
1007 if (i == 0) {
1008 /* rtx first */
1009 for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) {
1010 int len = strlen(rt->rtsp_streams[rtx]->control_url);
1011 if (len >= 4 &&
1012 !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
1013 "/rtx"))
1014 break;
1015 }
1016 if (rtx == rt->nb_rtsp_streams)
1017 return -1; /* no RTX found */
1018 rtsp_st = rt->rtsp_streams[rtx];
1019 } else
1020 rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];
1021 } else
1022 rtsp_st = rt->rtsp_streams[i];
1023
1024 /* RTP/UDP */
1025 if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
1026 char buf[256];
1027
1028 if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
1029 port = reply->transports[0].client_port_min;
1030 goto have_port;
1031 }
1032
1033 /* first try in specified port range */
1034 if (RTSP_RTP_PORT_MIN != 0) {
1035 while (j <= RTSP_RTP_PORT_MAX) {
1036 ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
1037 "?localport=%d", j);
1038 /* we will use two ports per rtp stream (rtp and rtcp) */
1039 j += 2;
1040 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
1041 goto rtp_opened;
1042 }
1043 }
1044
1045 #if 0
1046 /* then try on any port */
1047 if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
1048 err = AVERROR_INVALIDDATA;
1049 goto fail;
1050 }
1051 #endif
1052
1053 rtp_opened:
1054 port = rtp_get_local_port(rtsp_st->rtp_handle);
1055 have_port:
1056 snprintf(transport, sizeof(transport) - 1,
1057 "%s/UDP;", trans_pref);
1058 if (rt->server_type != RTSP_SERVER_REAL)
1059 av_strlcat(transport, "unicast;", sizeof(transport));
1060 av_strlcatf(transport, sizeof(transport),
1061 "client_port=%d", port);
1062 if (rt->transport == RTSP_TRANSPORT_RTP &&
1063 !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1064 av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1065 }
1066
1067 /* RTP/TCP */
1068 else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1069 /** For WMS streams, the application streams are only used for
1070 * UDP. When trying to set it up for TCP streams, the server
1071 * will return an error. Therefore, we skip those streams. */
1072 if (rt->server_type == RTSP_SERVER_WMS &&
1073 s->streams[rtsp_st->stream_index]->codec->codec_type ==
1074 AVMEDIA_TYPE_DATA)
1075 continue;
1076 snprintf(transport, sizeof(transport) - 1,
1077 "%s/TCP;", trans_pref);
1078 if (rt->server_type == RTSP_SERVER_WMS)
1079 av_strlcat(transport, "unicast;", sizeof(transport));
1080 av_strlcatf(transport, sizeof(transport),
1081 "interleaved=%d-%d",
1082 interleave, interleave + 1);
1083 interleave += 2;
1084 }
1085
1086 else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1087 snprintf(transport, sizeof(transport) - 1,
1088 "%s/UDP;multicast", trans_pref);
1089 }
1090 if (s->oformat) {
1091 av_strlcat(transport, ";mode=receive", sizeof(transport));
1092 } else if (rt->server_type == RTSP_SERVER_REAL ||
1093 rt->server_type == RTSP_SERVER_WMS)
1094 av_strlcat(transport, ";mode=play", sizeof(transport));
1095 snprintf(cmd, sizeof(cmd),
1096 "Transport: %s\r\n",
1097 transport);
1098 if (i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1099 char real_res[41], real_csum[9];
1100 ff_rdt_calc_response_and_checksum(real_res, real_csum,
1101 real_challenge);
1102 av_strlcatf(cmd, sizeof(cmd),
1103 "If-Match: %s\r\n"
1104 "RealChallenge2: %s, sd=%s\r\n",
1105 rt->session_id, real_res, real_csum);
1106 }
1107 ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1108 if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
1109 err = 1;
1110 goto fail;
1111 } else if (reply->status_code != RTSP_STATUS_OK ||
1112 reply->nb_transports != 1) {
1113 err = AVERROR_INVALIDDATA;
1114 goto fail;
1115 }
1116
1117 /* XXX: same protocol for all streams is required */
1118 if (i > 0) {
1119 if (reply->transports[0].lower_transport != rt->lower_transport ||
1120 reply->transports[0].transport != rt->transport) {
1121 err = AVERROR_INVALIDDATA;
1122 goto fail;
1123 }
1124 } else {
1125 rt->lower_transport = reply->transports[0].lower_transport;
1126 rt->transport = reply->transports[0].transport;
1127 }
1128
1129 /* close RTP connection if not choosen */
1130 if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
1131 (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
1132 url_close(rtsp_st->rtp_handle);
1133 rtsp_st->rtp_handle = NULL;
1134 }
1135
1136 switch(reply->transports[0].lower_transport) {
1137 case RTSP_LOWER_TRANSPORT_TCP:
1138 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
1139 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
1140 break;
1141
1142 case RTSP_LOWER_TRANSPORT_UDP: {
1143 char url[1024];
1144
1145 /* XXX: also use address if specified */
1146 ff_url_join(url, sizeof(url), "rtp", NULL, host,
1147 reply->transports[0].server_port_min, NULL);
1148 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
1149 rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1150 err = AVERROR_INVALIDDATA;
1151 goto fail;
1152 }
1153 /* Try to initialize the connection state in a
1154 * potential NAT router by sending dummy packets.
1155 * RTP/RTCP dummy packets are used for RDT, too.
1156 */
1157 if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat)
1158 rtp_send_punch_packets(rtsp_st->rtp_handle);
1159 break;
1160 }
1161 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
1162 char url[1024];
1163 struct in_addr in;
1164 int port, ttl;
1165
1166 if (reply->transports[0].destination) {
1167 in.s_addr = htonl(reply->transports[0].destination);
1168 port = reply->transports[0].port_min;
1169 ttl = reply->transports[0].ttl;
1170 } else {
1171 in = rtsp_st->sdp_ip;
1172 port = rtsp_st->sdp_port;
1173 ttl = rtsp_st->sdp_ttl;
1174 }
1175 ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(in),
1176 port, "?ttl=%d", ttl);
1177 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1178 err = AVERROR_INVALIDDATA;
1179 goto fail;
1180 }
1181 break;
1182 }
1183 }
1184
1185 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1186 goto fail;
1187 }
1188
1189 if (reply->timeout > 0)
1190 rt->timeout = reply->timeout;
1191
1192 if (rt->server_type == RTSP_SERVER_REAL)
1193 rt->need_subscription = 1;
1194
1195 return 0;
1196
1197 fail:
1198 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1199 if (rt->rtsp_streams[i]->rtp_handle) {
1200 url_close(rt->rtsp_streams[i]->rtp_handle);
1201 rt->rtsp_streams[i]->rtp_handle = NULL;
1202 }
1203 }
1204 return err;
1205 }
1206
1207 static int rtsp_read_play(AVFormatContext *s)
1208 {
1209 RTSPState *rt = s->priv_data;
1210 RTSPMessageHeader reply1, *reply = &reply1;
1211 int i;
1212 char cmd[1024];
1213
1214 av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
1215
1216 if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
1217 if (rt->state == RTSP_STATE_PAUSED) {
1218 cmd[0] = 0;
1219 } else {
1220 snprintf(cmd, sizeof(cmd),
1221 "Range: npt=%0.3f-\r\n",
1222 (double)rt->seek_timestamp / AV_TIME_BASE);
1223 }
1224 ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
1225 if (reply->status_code != RTSP_STATUS_OK) {
1226 return -1;
1227 }
1228 if (reply->range_start != AV_NOPTS_VALUE &&
1229 rt->transport == RTSP_TRANSPORT_RTP) {
1230 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1231 RTSPStream *rtsp_st = rt->rtsp_streams[i];
1232 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1233 AVStream *st = NULL;
1234 if (!rtpctx)
1235 continue;
1236 if (rtsp_st->stream_index >= 0)
1237 st = s->streams[rtsp_st->stream_index];
1238 rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE;
1239 rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE;
1240 if (st)
1241 rtpctx->range_start_offset = av_rescale_q(reply->range_start,
1242 AV_TIME_BASE_Q,
1243 st->time_base);
1244 }
1245 }
1246 }
1247 rt->state = RTSP_STATE_STREAMING;
1248 return 0;
1249 }
1250
1251 static int rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
1252 {
1253 RTSPState *rt = s->priv_data;
1254 char cmd[1024];
1255 unsigned char *content = NULL;
1256 int ret;
1257
1258 /* describe the stream */
1259 snprintf(cmd, sizeof(cmd),
1260 "Accept: application/sdp\r\n");
1261 if (rt->server_type == RTSP_SERVER_REAL) {
1262 /**
1263 * The Require: attribute is needed for proper streaming from
1264 * Realmedia servers.
1265 */
1266 av_strlcat(cmd,
1267 "Require: com.real.retain-entity-for-setup\r\n",
1268 sizeof(cmd));
1269 }
1270 ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content);
1271 if (!content)
1272 return AVERROR_INVALIDDATA;
1273 if (reply->status_code != RTSP_STATUS_OK) {
1274 av_freep(&content);
1275 return AVERROR_INVALIDDATA;
1276 }
1277
1278 /* now we got the SDP description, we parse it */
1279 ret = sdp_parse(s, (const char *)content);
1280 av_freep(&content);
1281 if (ret < 0)
1282 return AVERROR_INVALIDDATA;
1283
1284 return 0;
1285 }
1286
1287 static int rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
1288 {
1289 RTSPState *rt = s->priv_data;
1290 RTSPMessageHeader reply1, *reply = &reply1;
1291 int i;
1292 char *sdp;
1293 AVFormatContext sdp_ctx, *ctx_array[1];
1294
1295 rt->start_time = av_gettime();
1296
1297 /* Announce the stream */
1298 sdp = av_mallocz(8192);
1299 if (sdp == NULL)
1300 return AVERROR(ENOMEM);
1301 /* We create the SDP based on the RTSP AVFormatContext where we
1302 * aren't allowed to change the filename field. (We create the SDP
1303 * based on the RTSP context since the contexts for the RTP streams
1304 * don't exist yet.) In order to specify a custom URL with the actual
1305 * peer IP instead of the originally specified hostname, we create
1306 * a temporary copy of the AVFormatContext, where the custom URL is set.
1307 *
1308 * FIXME: Create the SDP without copying the AVFormatContext.
1309 * This either requires setting up the RTP stream AVFormatContexts
1310 * already here (complicating things immensely) or getting a more
1311 * flexible SDP creation interface.
1312 */
1313 sdp_ctx = *s;
1314 ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
1315 "rtsp", NULL, addr, -1, NULL);
1316 ctx_array[0] = &sdp_ctx;
1317 if (avf_sdp_create(ctx_array, 1, sdp, 8192)) {
1318 av_free(sdp);
1319 return AVERROR_INVALIDDATA;
1320 }
1321 av_log(s, AV_LOG_INFO, "SDP:\n%s\n", sdp);
1322 ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
1323 "Content-Type: application/sdp\r\n",
1324 reply, NULL, sdp, strlen(sdp));
1325 av_free(sdp);
1326 if (reply->status_code != RTSP_STATUS_OK)
1327 return AVERROR_INVALIDDATA;
1328
1329 /* Set up the RTSPStreams for each AVStream */
1330 for (i = 0; i < s->nb_streams; i++) {
1331 RTSPStream *rtsp_st;
1332 AVStream *st = s->streams[i];
1333
1334 rtsp_st = av_mallocz(sizeof(RTSPStream));
1335 if (!rtsp_st)
1336 return AVERROR(ENOMEM);
1337 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
1338
1339 st->priv_data = rtsp_st;
1340 rtsp_st->stream_index = i;
1341
1342 av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
1343 /* Note, this must match the relative uri set in the sdp content */
1344 av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
1345 "/streamid=%d", i);
1346 }
1347
1348 return 0;
1349 }
1350
1351 void ff_rtsp_close_connections(AVFormatContext *s)
1352 {
1353 RTSPState *rt = s->priv_data;
1354 if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out);
1355 url_close(rt->rtsp_hd);
1356 rt->rtsp_hd = rt->rtsp_hd_out = NULL;
1357 }
1358
1359 int ff_rtsp_connect(AVFormatContext *s)
1360 {
1361 RTSPState *rt = s->priv_data;
1362 char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
1363 char *option_list, *option, *filename;
1364 int port, err, tcp_fd;
1365 RTSPMessageHeader reply1 = {0}, *reply = &reply1;
1366 int lower_transport_mask = 0;
1367 char real_challenge[64];
1368 struct sockaddr_storage peer;
1369 socklen_t peer_len = sizeof(peer);
1370
1371 if (!ff_network_init())
1372 return AVERROR(EIO);
1373 redirect:
1374 rt->control_transport = RTSP_MODE_PLAIN;
1375 /* extract hostname and port */
1376 av_url_split(NULL, 0, auth, sizeof(auth),
1377 host, sizeof(host), &port, path, sizeof(path), s->filename);
1378 if (*auth) {
1379 av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1380 }
1381 if (port < 0)
1382 port = RTSP_DEFAULT_PORT;
1383
1384 /* search for options */
1385 option_list = strrchr(path, '?');
1386 if (option_list) {
1387 /* Strip out the RTSP specific options, write out the rest of
1388 * the options back into the same string. */
1389 filename = option_list;
1390 while (option_list) {
1391 /* move the option pointer */
1392 option = ++option_list;
1393 option_list = strchr(option_list, '&');
1394 if (option_list)
1395 *option_list = 0;
1396
1397 /* handle the options */
1398 if (!strcmp(option, "udp")) {
1399 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1400 } else if (!strcmp(option, "multicast")) {
1401 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1402 } else if (!strcmp(option, "tcp")) {
1403 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1404 } else if(!strcmp(option, "http")) {
1405 lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1406 rt->control_transport = RTSP_MODE_TUNNEL;
1407 } else {
1408 /* Write options back into the buffer, using memmove instead
1409 * of strcpy since the strings may overlap. */
1410 int len = strlen(option);
1411 memmove(++filename, option, len);
1412 filename += len;
1413 if (option_list) *filename = '&';
1414 }
1415 }
1416 *filename = 0;
1417 }
1418
1419 if (!lower_transport_mask)
1420 lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1421
1422 if (s->oformat) {
1423 /* Only UDP or TCP - UDP multicast isn't supported. */
1424 lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
1425 (1 << RTSP_LOWER_TRANSPORT_TCP);
1426 if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) {
1427 av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1428 "only UDP and TCP are supported for output.\n");
1429 err = AVERROR(EINVAL);
1430 goto fail;
1431 }
1432 }
1433
1434 /* Construct the URI used in request; this is similar to s->filename,
1435 * but with authentication credentials removed and RTSP specific options
1436 * stripped out. */
1437 ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
1438 host, port, "%s", path);
1439
1440 if (rt->control_transport == RTSP_MODE_TUNNEL) {
1441 /* set up initial handshake for tunneling */
1442 char httpname[1024];
1443 char sessioncookie[17];
1444 char headers[1024];
1445
1446 ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path);
1447 snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x",
1448 av_get_random_seed(), av_get_random_seed());
1449
1450 /* GET requests */
1451 if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) {
1452 err = AVERROR(EIO);
1453 goto fail;
1454 }
1455
1456 /* generate GET headers */
1457 snprintf(headers, sizeof(headers),
1458 "x-sessioncookie: %s\r\n"
1459 "Accept: application/x-rtsp-tunnelled\r\n"
1460 "Pragma: no-cache\r\n"
1461 "Cache-Control: no-cache\r\n",
1462 sessioncookie);
1463 ff_http_set_headers(rt->rtsp_hd, headers);
1464
1465 /* complete the connection */
1466 if (url_connect(rt->rtsp_hd)) {
1467 err = AVERROR(EIO);
1468 goto fail;
1469 }
1470
1471 /* POST requests */
1472 if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) {
1473 err = AVERROR(EIO);
1474 goto fail;
1475 }
1476
1477 /* generate POST headers */
1478 snprintf(headers, sizeof(headers),
1479 "x-sessioncookie: %s\r\n"
1480 "Content-Type: application/x-rtsp-tunnelled\r\n"
1481 "Pragma: no-cache\r\n"
1482 "Cache-Control: no-cache\r\n"
1483 "Content-Length: 32767\r\n"
1484 "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n",
1485 sessioncookie);
1486 ff_http_set_headers(rt->rtsp_hd_out, headers);
1487 ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0);
1488
1489 /* Initialize the authentication state for the POST session. The HTTP
1490 * protocol implementation doesn't properly handle multi-pass
1491 * authentication for POST requests, since it would require one of
1492 * the following:
1493 * - implementing Expect: 100-continue, which many HTTP servers
1494 * don't support anyway, even less the RTSP servers that do HTTP
1495 * tunneling
1496 * - sending the whole POST data until getting a 401 reply specifying
1497 * what authentication method to use, then resending all that data
1498 * - waiting for potential 401 replies directly after sending the
1499 * POST header (waiting for some unspecified time)
1500 * Therefore, we copy the full auth state, which works for both basic
1501 * and digest. (For digest, we would have to synchronize the nonce
1502 * count variable between the two sessions, if we'd do more requests
1503 * with the original session, though.)
1504 */
1505 ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd);
1506
1507 /* complete the connection */
1508 if (url_connect(rt->rtsp_hd_out)) {
1509 err = AVERROR(EIO);
1510 goto fail;
1511 }
1512 } else {
1513 /* open the tcp connection */
1514 ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1515 if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) {
1516 err = AVERROR(EIO);
1517 goto fail;
1518 }
1519 rt->rtsp_hd_out = rt->rtsp_hd;
1520 }
1521 rt->seq = 0;
1522
1523 tcp_fd = url_get_file_handle(rt->rtsp_hd);
1524 if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
1525 getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
1526 NULL, 0, NI_NUMERICHOST);
1527 }
1528
1529 /* request options supported by the server; this also detects server
1530 * type */
1531 for (rt->server_type = RTSP_SERVER_RTP;;) {
1532 cmd[0] = 0;
1533 if (rt->server_type == RTSP_SERVER_REAL)
1534 av_strlcat(cmd,
1535 /**
1536 * The following entries are required for proper
1537 * streaming from a Realmedia server. They are
1538 * interdependent in some way although we currently
1539 * don't quite understand how. Values were copied
1540 * from mplayer SVN r23589.
1541 * @param CompanyID is a 16-byte ID in base64
1542 * @param ClientChallenge is a 16-byte ID in hex
1543 */
1544 "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
1545 "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
1546 "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
1547 "GUID: 00000000-0000-0000-0000-000000000000\r\n",
1548 sizeof(cmd));
1549 ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1550 if (reply->status_code != RTSP_STATUS_OK) {
1551 err = AVERROR_INVALIDDATA;
1552 goto fail;
1553 }
1554
1555 /* detect server type if not standard-compliant RTP */
1556 if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
1557 rt->server_type = RTSP_SERVER_REAL;
1558 continue;
1559 } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
1560 rt->server_type = RTSP_SERVER_WMS;
1561 } else if (rt->server_type == RTSP_SERVER_REAL)
1562 strcpy(real_challenge, reply->real_challenge);
1563 break;
1564 }
1565
1566 if (s->iformat)
1567 err = rtsp_setup_input_streams(s, reply);
1568 else
1569 err = rtsp_setup_output_streams(s, host);
1570 if (err)
1571 goto fail;
1572
1573 do {
1574 int lower_transport = ff_log2_tab[lower_transport_mask &
1575 ~(lower_transport_mask - 1)];
1576
1577 err = make_setup_request(s, host, port, lower_transport,
1578 rt->server_type == RTSP_SERVER_REAL ?
1579 real_challenge : NULL);
1580 if (err < 0)
1581 goto fail;
1582 lower_transport_mask &= ~(1 << lower_transport);
1583 if (lower_transport_mask == 0 && err == 1) {
1584 err = FF_NETERROR(EPROTONOSUPPORT);
1585 goto fail;
1586 }
1587 } while (err);
1588
1589 rt->state = RTSP_STATE_IDLE;
1590 rt->seek_timestamp = 0; /* default is to start stream at position zero */
1591 return 0;
1592 fail:
1593 ff_rtsp_close_streams(s);
1594 ff_rtsp_close_connections(s);
1595 if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
1596 av_strlcpy(s->filename, reply->location, sizeof(s->filename));
1597 av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
1598 reply->status_code,
1599 s->filename);
1600 goto redirect;
1601 }
1602 ff_network_close();
1603 return err;
1604 }
1605 #endif
1606
1607 #if CONFIG_RTSP_DEMUXER
1608 static int rtsp_read_header(AVFormatContext *s,
1609 AVFormatParameters *ap)
1610 {
1611 int ret;
1612
1613 ret = ff_rtsp_connect(s);
1614 if (ret)
1615 return ret;
1616
1617 if (ap->initial_pause) {
1618 /* do not start immediately */
1619 } else {
1620 if (rtsp_read_play(s) < 0) {
1621 ff_rtsp_close_streams(s);
1622 ff_rtsp_close_connections(s);
1623 return AVERROR_INVALIDDATA;
1624 }
1625 }
1626
1627 return 0;
1628 }
1629
1630 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1631 uint8_t *buf, int buf_size)
1632 {
1633 RTSPState *rt = s->priv_data;
1634 RTSPStream *rtsp_st;
1635 fd_set rfds;
1636 int fd, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0;
1637 struct timeval tv;
1638
1639 for (;;) {
1640 if (url_interrupt_cb())
1641 return AVERROR(EINTR);
1642 FD_ZERO(&rfds);
1643 if (rt->rtsp_hd) {
1644 tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
1645 FD_SET(tcp_fd, &rfds);
1646 } else {
1647 fd_max = 0;
1648 tcp_fd = -1;
1649 }
1650 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1651 rtsp_st = rt->rtsp_streams[i];
1652 if (rtsp_st->rtp_handle) {
1653 /* currently, we cannot probe RTCP handle because of
1654 * blocking restrictions */
1655 fd = url_get_file_handle(rtsp_st->rtp_handle);
1656 if (fd > fd_max)
1657 fd_max = fd;
1658 FD_SET(fd, &rfds);
1659 }
1660 }
1661 tv.tv_sec = 0;
1662 tv.tv_usec = SELECT_TIMEOUT_MS * 1000;
1663 n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
1664 if (n > 0) {
1665 timeout_cnt = 0;
1666 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1667 rtsp_st = rt->rtsp_streams[i];
1668 if (rtsp_st->rtp_handle) {
1669 fd = url_get_file_handle(rtsp_st->rtp_handle);
1670 if (FD_ISSET(fd, &rfds)) {
1671 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1672 if (ret > 0) {
1673 *prtsp_st = rtsp_st;
1674 return ret;
1675 }
1676 }
1677 }
1678 }
1679 #if CONFIG_RTSP_DEMUXER
1680 if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
1681 RTSPMessageHeader reply;
1682
1683 ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
1684 if (ret < 0)
1685 return ret;
1686 /* XXX: parse message */
1687 if (rt->state != RTSP_STATE_STREAMING)
1688 return 0;
1689 }
1690 #endif
1691 } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
1692 return FF_NETERROR(ETIMEDOUT);
1693 } else if (n < 0 && errno != EINTR)
1694 return AVERROR(errno);
1695 }
1696 }
1697
1698 static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1699 uint8_t *buf, int buf_size)
1700 {
1701 RTSPState *rt = s->priv_data;
1702 int id, len, i, ret;
1703 RTSPStream *rtsp_st;
1704
1705 #ifdef DEBUG_RTP_TCP
1706 dprintf(s, "tcp_read_packet:\n");
1707 #endif
1708 redo:
1709 for (;;) {
1710 RTSPMessageHeader reply;
1711
1712 ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
1713 if (ret == -1)
1714 return -1;
1715 if (ret == 1) /* received '$' */
1716 break;
1717 /* XXX: parse message */
1718 if (rt->state != RTSP_STATE_STREAMING)
1719 return 0;
1720 }
1721 ret = url_read_complete(rt->rtsp_hd, buf, 3);
1722 if (ret != 3)
1723 return -1;
1724 id = buf[0];
1725 len = AV_RB16(buf + 1);
1726 #ifdef DEBUG_RTP_TCP
1727 dprintf(s, "id=%d len=%d\n", id, len);
1728 #endif
1729 if (len > buf_size || len < 12)
1730 goto redo;
1731 /* get the data */
1732 ret = url_read_complete(rt->rtsp_hd, buf, len);
1733 if (ret != len)
1734 return -1;
1735 if (rt->transport == RTSP_TRANSPORT_RDT &&
1736 ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0)
1737 return -1;
1738
1739 /* find the matching stream */
1740 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1741 rtsp_st = rt->rtsp_streams[i];
1742 if (id >= rtsp_st->interleaved_min &&
1743 id <= rtsp_st->interleaved_max)
1744 goto found;
1745 }
1746 goto redo;
1747 found:
1748 *prtsp_st = rtsp_st;
1749 return len;
1750 }
1751
1752 static int rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
1753 {
1754 RTSPState *rt = s->priv_data;
1755 int ret, len;
1756 uint8_t buf[10 * RTP_MAX_PACKET_LENGTH];
1757 RTSPStream *rtsp_st;
1758
1759 /* get next frames from the same RTP packet */
1760 if (rt->cur_transport_priv) {
1761 if (rt->transport == RTSP_TRANSPORT_RDT) {
1762 ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1763 } else
1764 ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1765 if (ret == 0) {
1766 rt->cur_transport_priv = NULL;
1767 return 0;
1768 } else if (ret == 1) {
1769 return 0;
1770 } else
1771 rt->cur_transport_priv = NULL;
1772 }
1773
1774 /* read next RTP packet */
1775 redo:
1776 switch(rt->lower_transport) {
1777 default:
1778 #if CONFIG_RTSP_DEMUXER
1779 case RTSP_LOWER_TRANSPORT_TCP:
1780 len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1781 break;
1782 #endif
1783 case RTSP_LOWER_TRANSPORT_UDP:
1784 case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
1785 len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1786 if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
1787 rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
1788 break;
1789 }
1790 if (len < 0)
1791 return len;
1792 if (len == 0)
1793 return AVERROR_EOF;
1794 if (rt->transport == RTSP_TRANSPORT_RDT) {
1795 ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
1796 } else {
1797 ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
1798 if (ret < 0) {
1799 /* Either bad packet, or a RTCP packet. Check if the
1800 * first_rtcp_ntp_time field was initialized. */
1801 RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
1802 if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) {
1803 /* first_rtcp_ntp_time has been initialized for this stream,
1804 * copy the same value to all other uninitialized streams,
1805 * in order to map their timestamp origin to the same ntp time
1806 * as this one. */
1807 int i;
1808 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1809 RTPDemuxContext *rtpctx2 = rtsp_st->transport_priv;
1810 if (rtpctx2 &&
1811 rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE)
1812 rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;
1813 }
1814 }
1815 }
1816 }
1817 if (ret < 0)
1818 goto redo;
1819 if (ret == 1)
1820 /* more packets may follow, so we save the RTP context */
1821 rt->cur_transport_priv = rtsp_st->transport_priv;
1822
1823 return ret;
1824 }
1825
1826 static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt)
1827 {
1828 RTSPState *rt = s->priv_data;
1829 int ret;
1830 RTSPMessageHeader reply1, *reply = &reply1;
1831 char cmd[1024];
1832
1833 if (rt->server_type == RTSP_SERVER_REAL) {
1834 int i;
1835 enum AVDiscard cache[MAX_STREAMS];
1836
1837 for (i = 0; i < s->nb_streams; i++)
1838 cache[i] = s->streams[i]->discard;
1839
1840 if (!rt->need_subscription) {
1841 if (memcmp (cache, rt->real_setup_cache,
1842 sizeof(enum AVDiscard) * s->nb_streams)) {
1843 snprintf(cmd, sizeof(cmd),
1844 "Unsubscribe: %s\r\n",
1845 rt->last_subscription);
1846 ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
1847 cmd, reply, NULL);
1848 if (reply->status_code != RTSP_STATUS_OK)
1849 return AVERROR_INVALIDDATA;
1850 rt->need_subscription = 1;
1851 }
1852 }
1853
1854 if (rt->need_subscription) {
1855 int r, rule_nr, first = 1;
1856
1857 memcpy(rt->real_setup_cache, cache,
1858 sizeof(enum AVDiscard) * s->nb_streams);
1859 rt->last_subscription[0] = 0;
1860
1861 snprintf(cmd, sizeof(cmd),
1862 "Subscribe: ");
1863 for (i = 0; i < rt->nb_rtsp_streams; i++) {
1864 rule_nr = 0;
1865 for (r = 0; r < s->nb_streams; r++) {
1866 if (s->streams[r]->priv_data == rt->rtsp_streams[i]) {
1867 if (s->streams[r]->discard != AVDISCARD_ALL) {
1868 if (!first)
1869 av_strlcat(rt->last_subscription, ",",
1870 sizeof(rt->last_subscription));
1871 ff_rdt_subscribe_rule(
1872 rt->last_subscription,
1873 sizeof(rt->last_subscription), i, rule_nr);
1874 first = 0;
1875 }
1876 rule_nr++;
1877 }
1878 }
1879 }
1880 av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
1881 ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
1882 cmd, reply, NULL);
1883 if (reply->status_code != RTSP_STATUS_OK)
1884 return AVERROR_INVALIDDATA;
1885 rt->need_subscription = 0;
1886
1887 if (rt->state == RTSP_STATE_STREAMING)
1888 rtsp_read_play (s);
1889 }
1890 }
1891
1892 ret = rtsp_fetch_packet(s, pkt);
1893 if (ret < 0)
1894 return ret;
1895
1896 /* send dummy request to keep TCP connection alive */
1897 if ((rt->server_type == RTSP_SERVER_WMS ||
1898 rt->server_type == RTSP_SERVER_REAL) &&
1899 (av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
1900 if (rt->server_type == RTSP_SERVER_WMS) {
1901 ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
1902 } else {
1903 ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
1904 }
1905 }
1906
1907 return 0;
1908 }
1909
1910 /* pause the stream */
1911 static int rtsp_read_pause(AVFormatContext *s)
1912 {
1913 RTSPState *rt = s->priv_data;
1914 RTSPMessageHeader reply1, *reply = &reply1;
1915
1916 if (rt->state != RTSP_STATE_STREAMING)
1917 return 0;
1918 else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
1919 ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL);
1920 if (reply->status_code != RTSP_STATUS_OK) {
1921 return -1;
1922 }
1923 }
1924 rt->state = RTSP_STATE_PAUSED;
1925 return 0;
1926 }
1927
1928 static int rtsp_read_seek(AVFormatContext *s, int stream_index,
1929 int64_t timestamp, int flags)
1930 {
1931 RTSPState *rt = s->priv_data;
1932
1933 rt->seek_timestamp = av_rescale_q(timestamp,
1934 s->streams[stream_index]->time_base,
1935 AV_TIME_BASE_Q);
1936 switch(rt->state) {
1937 default:
1938 case RTSP_STATE_IDLE:
1939 break;
1940 case RTSP_STATE_STREAMING:
1941 if (rtsp_read_pause(s) != 0)
1942 return -1;
1943 rt->state = RTSP_STATE_SEEKING;
1944 if (rtsp_read_play(s) != 0)
1945 return -1;
1946 break;
1947 case RTSP_STATE_PAUSED:
1948 rt->state = RTSP_STATE_IDLE;
1949 break;
1950 }
1951 return 0;
1952 }
1953
1954 static int rtsp_read_close(AVFormatContext *s)
1955 {
1956 RTSPState *rt = s->priv_data;
1957
1958 #if 0
1959 /* NOTE: it is valid to flush the buffer here */
1960 if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1961 url_fclose(&rt->rtsp_gb);
1962 }
1963 #endif
1964 ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
1965
1966 ff_rtsp_close_streams(s);
1967 ff_rtsp_close_connections(s);
1968 ff_network_close();
1969 return 0;
1970 }
1971
1972 AVInputFormat rtsp_demuxer = {
1973 "rtsp",
1974 NULL_IF_CONFIG_SMALL("RTSP input format"),
1975 sizeof(RTSPState),
1976 rtsp_probe,
1977 rtsp_read_header,
1978 rtsp_read_packet,
1979 rtsp_read_close,
1980 rtsp_read_seek,
1981 .flags = AVFMT_NOFILE,
1982 .read_play = rtsp_read_play,
1983 .read_pause = rtsp_read_pause,
1984 };
1985 #endif
1986
1987 static int sdp_probe(AVProbeData *p1)
1988 {
1989 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1990
1991 /* we look for a line beginning "c=IN IP4" */
1992 while (p < p_end && *p != '\0') {
1993 if (p + sizeof("c=IN IP4") - 1 < p_end &&
1994 av_strstart(p, "c=IN IP4", NULL))
1995 return AVPROBE_SCORE_MAX / 2;
1996
1997 while (p < p_end - 1 && *p != '\n') p++;
1998 if (++p >= p_end)
1999 break;
2000 if (*p == '\r')
2001 p++;
2002 }
2003 return 0;
2004 }
2005
2006 #define SDP_MAX_SIZE 8192
2007
2008 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
2009 {
2010 RTSPState *rt = s->priv_data;
2011 RTSPStream *rtsp_st;
2012 int size, i, err;
2013 char *content;
2014 char url[1024];
2015
2016 if (!ff_network_init())
2017 return AVERROR(EIO);
2018
2019 /* read the whole sdp file */
2020 /* XXX: better loading */
2021 content = av_malloc(SDP_MAX_SIZE);
2022 size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
2023 if (size <= 0) {
2024 av_free(content);
2025 return AVERROR_INVALIDDATA;
2026 }
2027 content[size] ='\0';
2028
2029 sdp_parse(s, content);
2030 av_free(content);
2031
2032 /* open each RTP stream */
2033 for (i = 0; i < rt->nb_rtsp_streams; i++) {
2034 rtsp_st = rt->rtsp_streams[i];
2035
2036 ff_url_join(url, sizeof(url), "rtp", NULL,
2037 inet_ntoa(rtsp_st->sdp_ip), rtsp_st->sdp_port,
2038 "?localport=%d&ttl=%d", rtsp_st->sdp_port,
2039 rtsp_st->sdp_ttl);
2040 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
2041 err = AVERROR_INVALIDDATA;
2042 goto fail;
2043 }
2044 if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
2045 goto fail;
2046 }
2047 return 0;
2048 fail:
2049 ff_rtsp_close_streams(s);
2050 ff_network_close();
2051 return err;
2052 }
2053
2054 static int sdp_read_close(AVFormatContext *s)
2055 {
2056 ff_rtsp_close_streams(s);
2057 ff_network_close();
2058 return 0;
2059 }
2060
2061 AVInputFormat sdp_demuxer = {
2062 "sdp",
2063 NULL_IF_CONFIG_SMALL("SDP"),
2064 sizeof(RTSPState),
2065 sdp_probe,
2066 sdp_read_header,
2067 rtsp_fetch_packet,
2068 sdp_read_close,
2069 };